• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.12
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• pp.106-113
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• 1997

While driving, the low frequency interior noise below 200Hz causes the main component that irritates the auditory acoustic sense. But these passive control methods bring out increment in cost and weight of the vehicle and result in low efficiency. Recently, various ANC(Active Noise Control) methos to suppress the low frequency noise began to launch into application. In this study, we implemented the active noise control system for passenger vehicle to cancel the engine booming noise using DSP-based control unit, 4 micorphones, and 2 speakers. We used MEFX-LMS (Multiple Error Filtered X-Least Mean Square) algorithm since it can be easily implemented in real time. Also, MEFX-RLS algorithm was taken to enhance the suppression of the harmonic components of the engine booming noise inspite of its computational complexity. The performance of two adaptive algorithms were analyzed with experimental resutls.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.11a
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• pp.489-496
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• 2008

This paper investigates active noise control of ducts using Filtered-x Least Mean Square (FXLMS) algorithms to reduce noise transmission. Single channel FXLMS (MFXLSM) and multiple channel FXLMS (MFXLMS) algorithms are used to implement the active control systems. The transmission loss is significantly increased by SFXLMS but the sound pressure level (SPL) at the upstream of the error sensor is increased while that of downstream is very low. This increase of the upstream SPL causes the duct wall to vibrate and so to radiate noise. To prevent the wall vibration generated by the sound field upstream, global sound field control is required. To reduce SPL globally along the duct, active noise control using MFXLMS is implemented. We can then obtained globally reduced SPL. It is found experimentally that the vibration level, and so the radiated noise level. can be reduced by the active noise control using MFXLMS.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.2
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• pp.95-102
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• 1996

This paper describes the realization of a rela-time adaptive acoustic echo canceller, which adopts a microprogramming method, for removing acoustical echoes in speakerphone systems using th eADSP-2101 microprocessor with a pipeline and modified harvard architecture. We apply the LMS (least mean square) algorithm to estimate the coefficients of a transversal FIR filter. For the acustic adaptive echo canceller, we propose a parallel operation programming to imrove algorithm execution speed and apply a nonlinear quantization to reduce the quantization error caused by large dynamic range of voice signal.

• Journal of the Korean Institute of Telematics and Electronics C
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• v.34C no.7
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• pp.61-69
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• 1997

With the vigorous researches in the character recognition, the need to recognize run-on multilingual handwritten characters is increasing to provide uses with more comfortable PUI(pen user interface) environments. In general, many intermediate word candidates word candidates are generated in run-on multilingual recognition because there is no information of ending position and linguistic kind of character. To remove unnecessary word candidates which are generated in run-on multilingual recognition, we classify them into two groups and select the best candidate among the word candidates in the group where the final characater is completed using 5 attributes. In this research, we propose a method in order to select the best one candidate. It is called WRM (Weighted ranking method). The weights are adaptively trained by LMS(Least mean square) learning rule. Results show that the abilities of decision makin gusing weights are much better than those not using weights.

• Journal of the Korean Society for Precision Engineering
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• v.18 no.9
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• pp.71-81
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• 2001

This paper analyzes the effectiveness of minimizing vibration and sound transmission on/through a thin rectangular plate by both feedback control and hybrid control which combines adaptive feedforward control with a feedback loop. An experimental system identification technique using the matrix-fractional curve-fitting of the frequency response data is introduced for complex shaped structures. This identification technique reduces the model order o the MIMO(Multi-Input Multi-Output) system which simplifies the practical implementation. The adaptive feedforward control uses a Multiple filtered-x LMS(Least Mean Square) algorithm and the feedback control uses a multivariable digital LQG(Linear Quadratic Gaussian) algorithm. Experimental results show that an effective reduction of sound transmission is achieved by the hybrid control scheme when both vibration and noise measurement signals are incorporated in the controller.

• Journal of the Korean Society for Nondestructive Testing
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• v.25 no.3
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• pp.189-200
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• 2005

Ultrasonic inspection methods are widely used for detecting flaws in materials. One of the more popular methods involves the extraction of an appropriate set of features followed by the use of a neural network for the classification of the signals in the feature space. This paper describes an approach which uses LMS method to determine the coordinates of the ultrasonic probe followed by the use of SAFT to estimate the location of the ultrasonic reflector The method is employed for classifying NDE signals from the steam generator tubes in a nuclear power plant. The classification results using this scheme for the ultrasonic signals from cracks and deposits within steam generator tubes are presented.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2200-2203
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• 2003

In an active noise control(ANC) system using the Filtered-X least mean square(LMS) algorithm, the online secondary path modeling method by exploiting a random noise generator is applied. This method is suitable for secondary path modeling. However, it is increased the residual error of the ANC system. In this paper, we presents an ANC system improved online secondary path modeling method which is modified Kuo and Zhang model that is the secondary path estimation by the additive noise. In addition, our proposed model is used that additive noise is transformed into the signal multiplied reference signal by gain control parameter and delayed.

• Journal of the Korean Society for Precision Engineering
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• v.26 no.12
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• pp.146-153
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• 2009

One of the most important issues in the wearable healthcare sensors is to minimize the motion artifacts in the vital signals for continuous monitoring. This paper presents a reflected type photoplethysmograph (PPG) sensor for monitoring heart rates at the artery of the wrist. Active noise cancellation algorithm was applied to compensate the distorted signals by motions with Least Mean Square (LMS) adaptive filter algorithms, using acceleration signals from a MEMS accelerometer. Experiments with a watch type PPG sensor were performed to validate the proposed algorithm during typical daily motions such as walking and running. The developed sensor is suitable for ubiquitous healthcare system and monitoring vital arterial signals during surgery.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.19 no.1
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• pp.24-34
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• 2009

This paper investigates active noise control of ducts using filtered-x least mean square(FXLMS) algorithms to reduce noise transmission. Single channel FXLMS(SFXLSM) and multiple channel FXLMS(MFXLMS) algorithms are used to implement the active control systems. The transmission loss is significantly increased by SFXLMS but the sound pressure level(SPL) at the upstream of the error sensor is increased while that of downstream is very low. This increase of the upstream SPL causes the duct wall to vibrate and so to radiate noise. To prevent the wall vibration generated by the sound field upstream, global sound field control is required. To reduce SPL globally along the duct, active noise control using MFXLMS is implemented. We can then be obtained globally reduced SPL. It is found experimentally that the vibration level, and so the radiated noise level, can be reduced by the active noise control using MFXLMS.

• Journal of the Korean Society of Fisheries and Ocean Technology
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• v.30 no.2
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• pp.125-131
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• 1994

This paper describes a method to measure the distortion level of loudspeaker using a LMS(Least Mean Square) adaptive filter. Conventional technique to measure the distortion level uses a band-pass filter with a sharp cut-off frequency characteristics. However. such the band-pass filter has a bed time response characteristics. On the other hand, the proposed method offers us an easy way to measure the specified harmonic distortion level with a small hardware. Moreover, our method is not affected by noise which has no correlation with the test signal, and the measurement can be carried out in a noisy environment. The effectiveness of the proposed method is confirmed by experiment using a loudspeaker in a noisy room.